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https://git.ffmpeg.org/ffmpeg.git
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avfilter/af_asoftclip: rewrite oversampling
Fixes most aliasing issues.
This commit is contained in:
Paul B Mahol
parent
3e127b595a
commit
94d4cc24c3
1
configure
vendored
1
configure
vendored
@ -3548,7 +3548,6 @@ afir_filter_select="rdft"
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ametadata_filter_deps="avformat"
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amovie_filter_deps="avcodec avformat"
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aresample_filter_deps="swresample"
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asoftclip_filter_deps="swresample"
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asr_filter_deps="pocketsphinx"
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ass_filter_deps="libass"
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atempo_filter_deps="avcodec"
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@ -21,11 +21,12 @@
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libswresample/swresample.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#define MAX_OVERSAMPLE 64
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enum ASoftClipTypes {
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ASC_HARD = -1,
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ASC_TANH,
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@ -39,6 +40,14 @@ enum ASoftClipTypes {
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NB_TYPES,
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};
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typedef struct Lowpass {
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float fb0, fb1, fb2;
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float fa0, fa1, fa2;
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double db0, db1, db2;
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double da0, da1, da2;
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} Lowpass;
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typedef struct ASoftClipContext {
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const AVClass *class;
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@ -49,10 +58,8 @@ typedef struct ASoftClipContext {
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double output;
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double param;
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SwrContext *up_ctx;
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SwrContext *down_ctx;
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AVFrame *frame;
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Lowpass lowpass[MAX_OVERSAMPLE];
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AVFrame *frame[2];
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void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
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int nb_samples, int channels, int start, int end);
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@ -60,7 +67,6 @@ typedef struct ASoftClipContext {
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#define OFFSET(x) offsetof(ASoftClipContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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#define F AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption asoftclip_options[] = {
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{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
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@ -76,7 +82,7 @@ static const AVOption asoftclip_options[] = {
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{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
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{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
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{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
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{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
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{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
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{ NULL }
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};
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@ -85,8 +91,7 @@ AVFILTER_DEFINE_CLASS(asoftclip);
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static int query_formats(AVFilterContext *ctx)
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{
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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@ -100,42 +105,103 @@ static int query_formats(AVFilterContext *ctx)
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return ff_set_common_all_samplerates(ctx);
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}
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static void get_lowpass(Lowpass *s,
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double frequency,
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double sample_rate)
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{
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double w0 = 2 * M_PI * frequency / sample_rate;
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double alpha = sin(w0) / (2 * 0.8);
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double factor;
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s->da0 = 1 + alpha;
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s->da1 = -2 * cos(w0);
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s->da2 = 1 - alpha;
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s->db0 = (1 - cos(w0)) / 2;
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s->db1 = 1 - cos(w0);
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s->db2 = (1 - cos(w0)) / 2;
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s->da1 /= s->da0;
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s->da2 /= s->da0;
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s->db0 /= s->da0;
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s->db1 /= s->da0;
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s->db2 /= s->da0;
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s->da0 /= s->da0;
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factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
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s->db0 *= factor;
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s->db1 *= factor;
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s->db2 *= factor;
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s->fa0 = s->da0;
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s->fa1 = s->da1;
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s->fa2 = s->da2;
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s->fb0 = s->db0;
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s->fb1 = s->db1;
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s->fb2 = s->db2;
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}
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static inline float run_lowpassf(const Lowpass *const s,
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float src, float *w)
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{
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float dst;
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dst = src * s->fb0 + w[0];
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w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
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w[1] = s->fb2 * src - s->fa2 * dst;
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return dst;
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}
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static void filter_flt(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels,
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int start, int end)
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{
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const int oversample = s->oversample;
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const int nb_osamples = nb_samples * oversample;
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const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
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float threshold = s->threshold;
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float gain = s->output * threshold;
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float factor = 1.f / threshold;
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float param = s->param;
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for (int c = start; c < end; c++) {
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float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
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const float *src = sptr[c];
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float *dst = dptr[c];
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for (int n = 0; n < nb_samples; n++) {
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dst[oversample * n] = src[n];
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for (int m = 1; m < oversample; m++)
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dst[oversample * n + m] = 0.f;
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}
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanhf(src[n] * factor * param);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = tanhf(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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float sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= 1.5f)
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dst[n] = FFSIGN(sample);
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@ -145,22 +211,22 @@ static void filter_flt(ASoftClipContext *s,
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++) {
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float sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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dst[n] = sample / (sqrtf(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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float sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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@ -170,8 +236,8 @@ static void filter_flt(ASoftClipContext *s,
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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float sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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@ -181,53 +247,86 @@ static void filter_flt(ASoftClipContext *s,
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = erff(src[n] * factor);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = erff(dst[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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av_assert0(0);
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}
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w = (float *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
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for (int n = 0; n < nb_samples; n++)
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dst[n] = dst[n * oversample] * scale;
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}
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}
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static inline double run_lowpassd(const Lowpass *const s,
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double src, double *w)
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{
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double dst;
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dst = src * s->db0 + w[0];
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w[0] = s->db1 * src + w[1] - s->da1 * dst;
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w[1] = s->db2 * src - s->da2 * dst;
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return dst;
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}
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static void filter_dbl(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels,
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int start, int end)
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{
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const int oversample = s->oversample;
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const int nb_osamples = nb_samples * oversample;
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const double scale = oversample > 1 ? oversample * 0.5 : 1.;
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double threshold = s->threshold;
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double gain = s->output * threshold;
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double factor = 1. / threshold;
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double param = s->param;
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for (int c = start; c < end; c++) {
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double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
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const double *src = sptr[c];
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double *dst = dptr[c];
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for (int n = 0; n < nb_samples; n++) {
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dst[oversample * n] = src[n];
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for (int m = 1; m < oversample; m++)
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dst[oversample * n + m] = 0.f;
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}
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = av_clipd(src[n] * factor, -1., 1.);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = av_clipd(dst[n] * factor, -1., 1.);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanh(src[n] * factor * param);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = tanh(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2. / M_PI * atan(src[n] * factor * param);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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double sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= 1.5)
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dst[n] = FFSIGN(sample);
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@ -237,22 +336,22 @@ static void filter_dbl(ASoftClipContext *s,
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++) {
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double sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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dst[n] = sample / (sqrt(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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double sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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@ -262,8 +361,8 @@ static void filter_dbl(ASoftClipContext *s,
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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double sample = src[n] * factor;
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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@ -273,14 +372,21 @@ static void filter_dbl(ASoftClipContext *s,
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = erf(src[n] * factor);
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = erf(dst[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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av_assert0(0);
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}
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w = (double *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
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for (int n = 0; n < nb_samples; n++)
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dst[n] = dst[n * oversample] * scale;
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}
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}
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@ -288,47 +394,21 @@ static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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ASoftClipContext *s = ctx->priv;
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int ret;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLT:
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case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
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case AV_SAMPLE_FMT_DBL:
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case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
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default: av_assert0(0);
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}
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if (s->oversample <= 1)
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return 0;
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s->up_ctx = swr_alloc();
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s->down_ctx = swr_alloc();
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if (!s->up_ctx || !s->down_ctx)
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s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
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s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
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if (!s->frame[0] || !s->frame[1])
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return AVERROR(ENOMEM);
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av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0);
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av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0);
|
||||
av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
|
||||
|
||||
av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0);
|
||||
av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0);
|
||||
av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
|
||||
|
||||
av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0);
|
||||
av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0);
|
||||
av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
|
||||
|
||||
av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0);
|
||||
av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0);
|
||||
av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
|
||||
|
||||
ret = swr_init(s->up_ctx);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
ret = swr_init(s->down_ctx);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
for (int i = 0; i < MAX_OVERSAMPLE; i++) {
|
||||
get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
@ -361,14 +441,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
||||
AVFilterContext *ctx = inlink->dst;
|
||||
ASoftClipContext *s = ctx->priv;
|
||||
AVFilterLink *outlink = ctx->outputs[0];
|
||||
int ret, nb_samples, channels;
|
||||
int nb_samples, channels;
|
||||
ThreadData td;
|
||||
AVFrame *out;
|
||||
|
||||
if (av_frame_is_writable(in)) {
|
||||
if (av_frame_is_writable(in) && s->oversample == 1) {
|
||||
out = in;
|
||||
} else {
|
||||
out = ff_get_audio_buffer(outlink, in->nb_samples);
|
||||
out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
|
||||
if (!out) {
|
||||
av_frame_free(&in);
|
||||
return AVERROR(ENOMEM);
|
||||
@ -376,72 +456,29 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
||||
av_frame_copy_props(out, in);
|
||||
}
|
||||
|
||||
if (av_sample_fmt_is_planar(in->format)) {
|
||||
nb_samples = in->nb_samples;
|
||||
channels = in->channels;
|
||||
} else {
|
||||
nb_samples = in->channels * in->nb_samples;
|
||||
channels = 1;
|
||||
}
|
||||
nb_samples = in->nb_samples;
|
||||
channels = in->channels;
|
||||
|
||||
if (s->oversample > 1) {
|
||||
s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
|
||||
if (!s->frame) {
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto fail;
|
||||
}
|
||||
|
||||
ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample,
|
||||
(const uint8_t **)in->extended_data, in->nb_samples);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
td.in = s->frame;
|
||||
td.out = s->frame;
|
||||
td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
|
||||
td.channels = channels;
|
||||
ff_filter_execute(ctx, filter_channels, &td, NULL,
|
||||
FFMIN(channels, ff_filter_get_nb_threads(ctx)));
|
||||
|
||||
ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
|
||||
(const uint8_t **)s->frame->extended_data, ret);
|
||||
if (ret < 0)
|
||||
goto fail;
|
||||
|
||||
if (out->pts)
|
||||
out->pts -= s->delay;
|
||||
s->delay += in->nb_samples - ret;
|
||||
out->nb_samples = ret;
|
||||
|
||||
av_frame_free(&s->frame);
|
||||
} else {
|
||||
td.in = in;
|
||||
td.out = out;
|
||||
td.nb_samples = nb_samples;
|
||||
td.channels = channels;
|
||||
ff_filter_execute(ctx, filter_channels, &td, NULL,
|
||||
FFMIN(channels, ff_filter_get_nb_threads(ctx)));
|
||||
}
|
||||
td.in = in;
|
||||
td.out = out;
|
||||
td.nb_samples = nb_samples;
|
||||
td.channels = channels;
|
||||
ff_filter_execute(ctx, filter_channels, &td, NULL,
|
||||
FFMIN(channels, ff_filter_get_nb_threads(ctx)));
|
||||
|
||||
if (out != in)
|
||||
av_frame_free(&in);
|
||||
|
||||
out->nb_samples /= s->oversample;
|
||||
return ff_filter_frame(outlink, out);
|
||||
fail:
|
||||
if (out != in)
|
||||
av_frame_free(&out);
|
||||
av_frame_free(&in);
|
||||
av_frame_free(&s->frame);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx)
|
||||
{
|
||||
ASoftClipContext *s = ctx->priv;
|
||||
|
||||
swr_free(&s->up_ctx);
|
||||
swr_free(&s->down_ctx);
|
||||
av_frame_free(&s->frame[0]);
|
||||
av_frame_free(&s->frame[1]);
|
||||
}
|
||||
|
||||
static const AVFilterPad inputs[] = {
|
||||
|
||||
Loading…
Reference in New Issue
Block a user